Overview

• SIP over WebSocket (use real SIP in your web apps)
• Audio/video calls (WebRTC), instant messaging and presence
• Lightweight! (~140KB)
• Easy to use and powerful user API
• Works with OverSIP, Kamailio and Asterisk servers (more info)
• Written by the authors of draft-ietf-sipcore-sip-websocket and OverSIP

Getting Started

The following simple JavaScript code creates a JsSIP User Agent instance and makes a SIP call:

// Create our JsSIP instance and run it:varconfiguration={'outbound_proxy_set': 'ws://sip-ws.example.com','uri': 'sip:alice@example.com','password': 'superpassword'};varcoolPhone=newJsSIP.UA(configuration);coolPhone.start();// Make an audio/video call:varuseAudio=true;varuseVideo=true;// id attribute of existing HTML5 <video> elements in which local and remote video will be shownvarviews={'localView': "my-cam",'remoteView': "peer-cam"};vareventHandlers={'connecting': function(e){// Your code here },'progress': function(e){// Your code here },'failed': function(e){// Your code here },'started': function(e){// Your code here },'ended': function(e){// Your code here }};coolPhone.call('sip:bob@example.com',useAudio,useVideo,eventHandlers,views);

Want to see more? Check the full Getting Started section in the project website and our nice demos.

Website and Documentation

• jssip.net

Download

• jssip.net/download

Authors

Main Author

• José Luis Millán (jmillan@aliax.net | github | twitter)

Contributors

• Iñaki Baz Castillo (ibc@aliax.net | github | twitter)
• Saúl Ibarra Corretgé (saghul@gmail.com | github | twitter)

License

JsSIP is released under the MIT license.